Posts by Per
-
-
Stacks, amps and cabs... Is it means to be possible to swap these out, or are they all just a part of a single rig that you must find and duplicate each time you want to make a new preset based on that amp? Whenever i go to browse with them the presets list just contains about 4 items and I can't find a way to add any more (I'd have assumed that the amps and cabs from any rig loaded would have been added to a central list here, but that doesn't appear to be the case).
Also is it normal that in Browser mode the Browse wheel doesn't do anything and you need to click through rigs with the rig buttons? (if so then that seems like something that you'd want to change).
Is the EXIT Led ever meant to be on fully bright, or is it normal to only ever half light up?
Is there any way to adjust the tonality of the sound as it cleans up? I know there's "Sens" for the volume, but no "Tone Sens" as far as I can see. I'm having difficulty profiling the actual distortion of my Mesa except at high gain, the cleaning up and touch characteristic veers sharply away from the way the amp itself is (is there a way to get it to do a more detailed profiling at lower volumes perhaps?).
Thanks
-
LOL, in Greece we say "wanting the Pita (something like a cake I suppose) whole, and the dog not hungry".
The English is "having your cake and eating it", not a bad thing to wish for however I honestly didn't think it was too much to ask of an amp sim with two USB ports (obviously wasting space) on the back of it.
-
The process is as loud as you turn up your amp. All that needs to happen is that the KPA gets a good enough signal. So you can either use a passive mic like an SM57/58 and turn up your amp real loud. Or you could use a mic pre and then profile at a lower volume. I didn't find the profiling to be that much louder than playing guitar through the amp, so don't be too scared. Use the volume that you'd normally record at.
-
I think they need to start out by fixing the knobs so that they don't cause sudden massive jumps to start out with. But once that's done then I don't see much harm in "knob acceleration" provided it didn't happen too soon. An alternative might be to hold down the quick button and turn a knob to have it move much faster (or maybe some other button that's not in use at the same time as the knobs), or perhaps slower instead for fine tuning.
-
Display More
Oversampling two signals prior to summing and downsampling it afterwards will give the same results as when you just sum the two plain signals.
In other words: the glue that you produce by oversampling will be summed, but later discarded by downsampling, without having an effect to the residual signal. You don't gain anything by that.Where the hell have your found this theory?
You will not find any skilled DSP programmer stating that summing signals is sort of a challenge.
Please tell me the source where this kind of digital esoterik is teached.Digital summing is:
Have sample streams of the same sample rate.
You add every corresponding sample.
Especially when you mix many streams (mixing desk), maintain a high bit width until the last stream is added.
Then the result can be truncated (cut the bits not needed).CK
Hi Christoph, thanks for chiming in. So I was wrong then with audio. Thanks for clearing that up. However I do think that this discussion is a diversion. As originally requested I'd still like 48k or other sample rates from the S/PDIF (or to allow the convertor to set the world clock/master rather than the Kemper), if nothing else I don't see it as in any way being a negative for the unit and it may even help bring in more sales (or at least not turn off anyone who may have been interested otherwise).
-
What I said is not the same as you mentioned on your post or at least not the way you wrote it. I am talking about using signals originally sampled at a higher rate, then summing and downsampling at the end. That normally produces better results but just because the source you are using is better/more accurate.
Maybe you were just confusing digital summing vs digital to
analog (mixer) summing. Some people prefer doing the summing in analog
even if all the source tracks are in digital because it sounds better for them.
No, that is what I wrote. i.e. Making the assumption (which as it turns out is wrong - see Christoph's post above yours) two approaches, high sample rate or reconstruction before modification of the sound at a higher rate (so effectively both = higher sample rate).Anyhow based on the Nyquist rule (which is just a math rule, not specific to audio) then you'd need a sample rate of only above 2x the highest frequency that humans can hear in order to accurately reconstruct the waveform at the other end when accuracy = the same up to the limit of human hearing. So including what Christoph said above then unless higher frequencies/harmonics somehow were adding interference that's audible lower down (unlikely) then 44.1k should be just as "good" as 48k or 96k or even 192k, at least when it comes to human hearing, and the final result should null exactly if you were to take a mix made at 44.1k and one made at 192k and they were made to match frequency with the same aliasing filter as applied to the original sources for the 44.1k version (and according to what Christoph said that's regardless of whether they were reconstructed and re-aliased around the mixer/fx or if the samples went straight through the whole thing), and of course ignoring clock jitter. And I've heard that said before too elsewhere in these discussions.
I still prefer to work at 48k though, maybe it's purely psychosomatic, but it does feel easier to mix in and maintain clarity, or "better" as you say. Christoph has said in the past that there are parts of the KPA where the sampling rate is much higher, I wonder what they are and why they are there, also why audio software and interfaces offer higher samplerates at all, except maybe the cynic in me says as a meaningless checkbox to attract customers.
-
Yes that's what I meant (see my second post). And you may be right, but only Christoph can answer that. I'd be very surprised if the DSP couldn't handle it though given that even a Pod can do up to 96k and that's got a far less beefy DSP, also Christoph has said that internally it's working with much higher rates in certain places. But as mentioned, only Christoph knows if he can squeeze out that little bit more or not.
It would be sad if it's locked to 44.1 for it's lifetime, it would have been fun to use the Apogee convertors and keep an all digital signal path (not to mention less cables and therefore less hotswapping in order to use it), but being honest I don't hold much hope.
-
Yes it's vital. Fingers crossed that Kemper also thinks so.
-
No, you don't need it, but it sure would be nice to have. I've no idea how heave the DSP usage would be to do it and how much latency it would add, with the chip in the KPA and a nice algorithm it may be negligible, or it might be untenable, but there's no reason not to ask for it. The main idea though isn't so much to have convolution reverb itself as to be able to profile the room! And then the cool things you could do with that either during profiling of the amp or after as an effect.
-
Sorry, not aliasing filter (although that can involve a reconstruction algorithm), and not *just* the reconstruction filter either. You got me just as I was waking up and brain wasn't firing on all four. Anyhow. My suspicion was just that it's the combination of reconstructionon DAC and aliasing filter on ADC with the analog desk. And of course working at a higher bitrate cuts out the middle man so to speak, thus similar "glue", as obviously you'd get some interference down to an octave lower than your highest point when summing without that on final DAC.
Anyhow, you know all this stuff I'm sure.
-
In audio it'd be an aliasing filter. It's the same Reconstruction that goes on in a DAC. There's some explanation here : http://en.wikipedia.org/wiki/Reconstruction_filter
You need this during summing, or really most stuff modifying audio, otherwise the nyquist shannon rule simply doesn't apply, after all it's a rule about integration/reconstruction itself. Anyhow you can end up with... I guess they'd be moire patterns If it were graphics, but odd interference patterns with the result, possible subtle audio artifacts. As DAC handles this then summing through a nice analog mixing desk means 44.1 is plenty, and those that do tend to talk about diffuse terms such as "glue", I suspect it has less to do with the analog components, and more to do with the reconstruction filter before summing and resampling with aliasing after summing than any coloration added between the two. People who work at higher sample rates also tend to talk about this "glue" phenomena while mixing, and personally speaking while I definitely don't have golden ears, there does seem to be something slightly easier about mixing with audio at a higher bit rate. It's just a hunch that this is the reason.
-
Sounds like a jolly good idea to me. I think it could work just fine, if the kemper dropped down a single loud "click" at the beginning then you could record the result, jam it on the usb thumbdrive and the KPA would be able to work out how to line up the results. To allow refining the KPA would have to add another click marker so it could find and align up the refining notes you play with the result on the amp. You'd need both the original source file and the re-amped file though for it to work.
-
Thought I'd give this profiling lark a quick try. I'll talk about my thoughts on the profiling, accuracy etc elsewhere. Lets just say that the results are at least interesting and fun in their own right, even with a crappy signal chain.
Speaking of which, this used a bog standard Shure SM58/7 through a nasty little Presonus tube-pre that I had laying around. The Mesa itself runs through a THD hotplate and you're listening to the burn channel. The "FAR" version has the mix placed a couple of meters back from the amp, the other is on axis off center of cone, to my ears incredibly fizzy and brittle sounding in the profile (my recording experience with this amp goes quite counter to that, it tends to suffer more from flubbyitis). Apologies about the tagging/naming, I thought it would carry over my name as I'd already entered it into one. Guitar is a Parker P-38 neck pickup (single coil).
The demo improv clip is using the far version (bass is through the same profile with the gain dialed back a little).
http://www.peranders.com/music/music/noodle120212a.mp3
Here's the kippers :
http://www.per-anders.net/gene…45%20-%20MESA%20BURN.kipr
http://www.per-anders.net/gene…%20MESA%20BURN%20FAR.kipr -
Display More
I'm an IP attorney (trademarks generally) and what you say is not necessarily true regarding the patent. It really goes to how well the patent claims were drafted in the patent application and whether the implementation used by the "alleged infringing party" overlaps those claims. There is more than one way to build a mouse-trap.
Also, sometime "winning" the lawsuit is not the objective, but having the financial resources to sue and forcing a less financially capable company from continuing using the tech or forcing them to license the tech. License fees can be quite lucrative.
In terms of Line6 not suing anybody, I remember checking a few years ago and not finding any case, but I believe Behringer licensed the Pod technology for the V-amp.
Finally, calling the tech by a different name is a trademark issue, not patent issue.
I love that music brings together so many people from so many different backgrounds. Anyhow, what you say reaffirms my thoughts on reading this, but I have to ask, doesn't that also make it a bit of a double edged sword? -
An amp invites you to control?
A guitar invites you to master?
Q. Does a profile of a modeler have Vai nature or not? A. Wú.
-
RME all have spdif i/o, they're just not always via coax. The Babyface uses dual use ADAT ports. You can have either spdif i/o or 8 ADAT i/o. All that's needed are a couple of dirt cheap cable converters from ADAT to coax.
http://www.rme-audio.de/en_products_babyface.php
Can't fault it. My third RME.
You're right. My bad. I do think that (outside of the pre's) the RME is a much better option than the Apogee, they really pack in the features, have great support and of course maintain a product that's viable on all platforms and isn't trying to just be a cool looking gimmick. -
Thank you professor for enlightening my path.I disagree with every single thing you've written (and that's included the full stops you've used ). Also when you say "Apogee or RME converters dont actually have SPDIF inputs " you clearly show you don't know what you're talking about.
And when you say "the UFX for instance has not just inboard effects, but an inbuilt sampler!" . Seriuosly I really I think you've got a talent ! Don't waste it in here!You're way ahead man
OK, editing this down to less waffle and more meat.1) I have an Ensemble, it has both Coax and Optical S/PDIF
2) Please don't paraphrase to purposefully misquote me, either that or reading comprehension 101 for you. Most guitarists I know use cheaper units that do not have S/PDIF, that's quite common even among the low end offerings from the big boys, in fact most frequently...
3) ...Guitarists these days expect their multiFX to be able to record over USB and just use that. I'm having a hard time thinking of one outside of the Kemper that doesn't.
4) And in fiscal terms you can't depend on studios to sell the Kemper. They're going out of business! The biggest slice of the pie is the hobby market, i.e. those guitarists. I have to question your business acumen based on your assertions and assumptions.
5) The UFX ok, not technically a sampler, but DURec direct recording to an attached drive for retroactive takes which is a lovely feature even if some DAWs already support this ITB. RME's have for a long time had onboard FX. By your standards these are bloatware features, personally I think they're excellent additions.
6) The Kemper is going to have to get USB drivers of some sort if you want even a basic librarian that doesn't involve constantly plugging and unplugging a usb thumbdrive between computer and KPA. I doubt that for someone who has a history of making USB audio with the Virus TI an ASIO drive and it's support is any worse than any other key feature that needs to be added.
7) USB audio support has had no noticeable impact on AxeFX development progress, do you not agree? To me it sounds like you think Christoph isn't up to the job. At least that's the implication. That's pretty insulting.So, do you have a comment to make or are you just being obnoxious for the sake of it?
-
Would be real nice, especially if we could use the KPA to generate the IR for it too. Then you'd be able to profile not just the amp but also the room! You could even have a dial to add/remove more or less room from the profile once you'd grabbed it, a bit like is done when recording audio on set.
-
Apogee is not a good example. OSX only in an era when even apple computers are dual boot, not to mention their continuing drive away from pro audio into iPad trinkets and stuff designed to be sold at apple stores first and foremost. And this is coming from an Apogee user.
RME also don't make a good example as their units are increasingly bloating away from just being dedicated converters, the UFX for instance has not just inboard effects, but an inbuilt sampler!
If you then factor in the cost, and the pain of SPDIF, especially if you have other SPDIF equipment so you need to daisy chain, and the fact that most guitarists, even the ones with Apogee or RME converters dont actually have SPDIF inputs (theyRe not on Babyface, Duet, MBox Mini etc) then it really does make more sense to allow normal guitarists who don't own a studio to be allowed to use USB in order to quickly and conveniently track their stuff with a digital signal path.
